date:20051003

This is a wierd one.  Can't figure it out.  I have an SPA-3000 at the
house handling my incoming line.  It's setup to direct the incoming call
to asterisk.  Works great 99% of the time.

A few times a day, I'm getting calls that ring once internally and are
then hungup.  I managed to get a detailed log [1] of what's happening
today and it looks to me that the SPA is acting wierd.  Can someone verify
this for me?

I looks to me that the Sipura is just CANCEL'ing the call shortly (2 secs
in this example) after setting it up.  I'm looking for someone to verify
this before I stop looking at Asterisk as the cause and focus on the SPA.

Thanks in advance,

Paul

[1] http://dugas.cc/~pdugas/spa3k.log
-- 
Paul Dugas, Computer Engineer   Dugas Enterprises, LLC
[EMAIL PROTECTED] phone: 404-932-1355   522 Black Canyon Park
http://dugas.cc fax: 866-751-6494   Canton, GA 30114 USA
--
Onsite at GDOT W.Annex 404-463-2860 x199
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Re: [Asterisk-Users] Music on hold not initiating RTP stream?

2005-10-03 Thread Michael George

On Fri, Sep 30, 2005 at 06:47:51PM -0500, Kevin P. Fleming wrote:
 Ray Van Dolson wrote:
 
 The ATA's are Sipura SPA-2002's and I have MOH Server set to 899 on each.
 
 Take that out, you don't need it.

He had this in there for testing to show that the problem was not mpg123,
which he did.

 However, with a call in progress, if I hit hold or flash on SIP ATA 1, it
 behaves correctly, but no music on hold is heard on SIP ATA 2.  I can see 
 in
 my Asterisk console that MusicOnHold() gets called and tcpdump shows the
 INVITE that first sets the RTP source to 0.0.0.0 then sets it to the IP of 
 my
 Asterisk box.
 
 None of this is needed; Asterisk will stream MOH to ATA 2 all by itself, 
 just by the fact that ATA 1 put ATA 2 on hold. You have 
 over-complexified the setup :-)

I'm not sure what you mean here.  You do have to defind a MOH class for any
channel not using default.

I think the problem you have is that you have not indicated anywhere that you
have set the MOH class for either channel to random.  If you do not do that,
it will try to use MOH class default. 

Make sure you test the default class with your 899 extension, or set the MOH
class to random for the channels you are testing.

HTH.

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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RE: [Asterisk-Users] X100p Problem, randomly hungup pstn line

2005-10-03 Thread Faris Raouf

I installed this card, everything work, i can make call and receive
call with no echo and great sound quality, but after between 5 to 50
secs the call disconnect by itself, in the log i don't see nothing
revelant.

In logging.conf, try enabling debug logging to the console and/or to
/var/log/asterisk/messages to see if you can find the cause. chan_zap.c
displays a lot of useful debug info if you enable the debug level logging.

Also please post your zaptel.conf and zapata.conf config so we can have a
look.

Faris.


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[Asterisk-Users] How to establish ISDN port Up

2005-10-03 Thread Damian Minkov


  We have ISDN PCI Adapter from Billion(using mISDN). When we connect it to the
PSTN the incoming calls are OK. When we try to make a call we can't do
that because the ISDN port is Down:
 CLI misdn show port 1
 BEGIN STACK_LIST:
 * Stack Addr: 4041 Port 1 Type TE Prot. PMP L2Link DOWN
 L1Link:DOWN
 Idx: 0 stack-chan: 0 Chan 1 InUse:0
 Idx: 1 stack-chan: 0 Chan 2 InUse:0

  When we try to up manually we have the following:
 misdn port up 1
 NO BC FOR STACK: port:1
 TE_FRM_HANDLER: Returning 0 on prim:20382 port:1
 Unhandled Message:
prim 20382 len 0 from addr 4141, dinfo 2 on port: 1


  How can estabilish the ISDN port to Up ?




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[Asterisk-Users] [Fwd: Eicon Diva 2.01 S/T PCI quality problems]

2005-10-03 Thread Kristof Jozsa


Hi all,

I'm experimenting with Asterisk on linux using an Eicon Diva 2.01 S/T PCI card.
I'd set up the card using the hisax driver and isdn4linux (titled as Old
ISDN4Linux (obsolete) in the 2.6.12 kernel. I can make SIP calls and outgoing
phone calls as well, but gathered a few problems on my way:

1. Plain SIP calls using softphones on windows clients work fine not counting
the delay I'm experimenting. We talk about 1-1.5 secs delay in the speech which
is rather distrubing (no noise in the line though).

2. Outgoing calls eg. to my mobile phone has some more serious problems. Speech
quality on my mobile phone is excellent. However, sound quality on the asterisk
console machine where I dialled from is about unacceptable. It has about 90%
static noise and about 10% speech somewhere in the middle. I have the same
experience calling from a windows softphone through asterisk to my handy, maybe
a little less noise (around 80%).

So my questions would be: can I do anything with the issues described above? Is
it a hardware problem (eg. I need to replace the old and cheap card to a more
modern one)? Maybe it's a driver problem (eg. the hisax driver is known to work
only with such extreme static noise)? I also don't really understand why the
static appears only on the server side and not on the called side.

Any help or suggestions are much appreciated,
thanks very much in advance.

Kristof Jozsa

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Re: [Asterisk-Users] SPA-3000 generating one-ring calls

Do you have the SIP acct it's interacting with enabled for MWI? There's a setting in the SPA3k where it will ring the phone periodically for one ring in addition to the stutter tone for MWI.
On 10/3/05, Paul Dugas [EMAIL PROTECTED] wrote:
This is a wierd one.Can't figure it out.I have an SPA-3000 at thehouse handling my incoming line.It's setup to direct the incoming call
to asterisk.Works great 99% of the time.A few times a day, I'm getting calls that ring once internally and arethen hungup.I managed to get a detailed log [1] of what's happeningtoday and it looks to me that the SPA is acting wierd.Can someone verify
this for me?I looks to me that the Sipura is just CANCEL'ing the call shortly (2 secsin this example) after setting it up.I'm looking for someone to verifythis before I stop looking at Asterisk as the cause and focus on the SPA.
Thanks in advance,Paul[1] http://dugas.cc/~pdugas/spa3k.log--Paul Dugas, Computer Engineer Dugas Enterprises, LLC
[EMAIL PROTECTED] phone: 404-932-1355 522 Black Canyon Parkhttp://dugas.cc fax: 866-751-6494 Canton, GA 30114 USA--Onsite at GDOT W.Annex 404-463-2860 x199___
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RE: [Asterisk-Users] codec g723 on Via C3

2005-10-03 Thread Juan Salas




I have 
a VIA Samuel 2, I use the Intel primitives (g729)
setting the Makefile to a 586 processor.
Maybe 
you can test with this.

Regards.

Jsalas.

  -Mensaje original-De: Giordano Grandis 
  [mailto:[EMAIL PROTECTED]Enviado el: Monday, October 03, 2005 
  7:06 AMPara: Asterisk Users Mailing List - Non-Commercial 
  DiscussionAsunto: [Asterisk-Users] codec g723 on Via 
  C3
  
  Hi,
  just a question: anyone has never 
  installed g729 codec on VIA motherboard with C3 processor 
  ?
  
  I'm having problem with IPP 
  libraries, and Intel said that it works only on Inter 
  processor.
  
  Any 
  suggestion?
  
  Thanks
  
  Giordano
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[Asterisk-Users] fc4 + iax + trunking

2005-10-03 Thread Raul Elizondo $wizardteam$

Hi,

I m using asterisk on a system without a digium hardware, and when i try to
use trunk=yes for my iax2 links, i just get this message while debugging:

Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
   Timestamp: 6ms  SCall: 16384  DCall: 16384 [192.168.1.1:4569]
   AUTHMETHODS : 2
   CHALLENGE   : 753310823
   USERNAME: 350001

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL
   Timestamp: 0ms  SCall: 16384  DCall: 16384 [192.168.1.1:4569]

if i remove trunk=yes, it works, but i would like to run multiple channels
at the same time.  zaptel was compiled normal (make clean ; make linux26 ;
make install) and ztdummy was loaded without problems (modprobe ztdummy)
which also loads zaptel driver.

Does anyone has this problem too and a hint to fix it?

Regards,

-=Raul=-


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Re: [Asterisk-Users] SPA-3000 generating one-ring calls

On Mon, October 3, 2005 9:33 am, BJ Weschke wrote:
 Do you have the SIP acct it's interacting with enabled for MWI? There's a
 setting in the SPA3k where it will ring the phone periodically for one
 ring in addition to the stutter tone for MWI.

I'm not using the SPA3k as an extension at the moment; just as an FXO
interface.  The SPA is initiating a SIP call to the Asterisk server then
DELETE'ing it 2 secs later.  Asterisk is ringing other IAX/SIP extensions
in response.  The FXS interface of the SPA3k *is* setup and registering
with the server but it's never getting called and it doesn't have anything
connected to it.

Curious,

Paul
-- 
Paul Dugas, Computer Engineer   Dugas Enterprises, LLC
[EMAIL PROTECTED] phone: 404-932-1355   522 Black Canyon Park
http://dugas.cc fax: 866-751-6494   Canton, GA 30114 USA
--
Onsite at GDOT W.Annex 404-463-2860 x199
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[Asterisk-Users] Problem with configuration of Quintum AX with Asterisk

2005-10-03 Thread Michal Misiak

Hi. 
I'm trying to configurate Quintum AX to work with Asterisk SIP egister/Proxy
server and my problem is that only the first user account get logged in and
only that user is able to make call correctly. It seems to be a problem with
authorization - I have noticed no Proxy-Authorization information in SIP
INVITE, ACK, CANCEL messages. I have also noticed that when I remove secret
from Asterisk sip.conf file (no authorization required) other users
(accounts) can make a call but no media are sent. 

Do you know reasons of this problem and can you help me resolving it. 


Michał Misiak
--
Have a nice day!
phone: (+48 22) 4330419
mobile: (+48) 888 395 336
e-mail: [EMAIL PROTECTED]
homepage: www.michalmisiak.prv.pl


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R: [Asterisk-Users] codec g723 on Via C3

2005-10-03 Thread Giordano Grandis









Thankswhich
version of IPP did u use ?

I do not have Makefile
file.there is only a .sh script



Thanks





Giordano











Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Juan Salas
Inviato: lunedì 3 ottobre 2005
15.41
A: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Oggetto: RE: [Asterisk-Users]
codec g723 on Via C3







I have a VIA Samuel 2, I use the Intel
primitives (g729)





setting the Makefile to a 586 processor.





Maybe you can test with this.











Regards.











Jsalas.





-Mensaje original-
De: Giordano Grandis
[mailto:[EMAIL PROTECTED]
Enviado el: Monday, October 03,
2005 7:06 AM
Para: Asterisk Users Mailing List
- Non-Commercial Discussion
Asunto: [Asterisk-Users] codec
g723 on Via C3

Hi,

just a question: anyone has never installed g729
codec on VIA motherboard with C3 processor ?



I'm having problem with IPP libraries, and Intel said
that it works only on Inter processor.



Any suggestion?



Thanks



Giordano








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Re: [Asterisk-Users] SPA-3000 generating one-ring calls

What is the FXO port connected to? An ATA from another VoIP provider? I've seen this same behavior when I reset the ATA that I have for ATT CallVantage service at home that is connected to the FXO port of a spa3k. I've got to imagine it is some kind of momentary dip or spike in the line voltage that is coming through the FXO port. 

On 10/3/05, Paul Dugas [EMAIL PROTECTED] wrote:
On Mon, October 3, 2005 9:33 am, BJ Weschke wrote: Do you have the SIP acct it's interacting with enabled for MWI? There's a
 setting in the SPA3k where it will ring the phone periodically for one ring in addition to the stutter tone for MWI.I'm not using the SPA3k as an extension at the moment; just as an FXOinterface.The SPA is initiating a SIP call to the Asterisk server then
DELETE'ing it 2 secs later.Asterisk is ringing other IAX/SIP extensionsin response.The FXS interface of the SPA3k *is* setup and registeringwith the server but it's never getting called and it doesn't have anything
connected to it.Curious,Paul--Paul Dugas, Computer Engineer Dugas Enterprises, LLC[EMAIL PROTECTED] phone: 404-932-1355 522 Black Canyon Park
http://dugas.cc fax: 866-751-6494 Canton, GA 30114 USA--Onsite at GDOT W.Annex 404-463-2860 x199
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Re: [Asterisk-Users] SPA-3000 generating one-ring calls

On Mon, October 3, 2005 10:10 am, BJ Weschke wrote:
 What is the FXO port connected to? An ATA from another VoIP provider?

It's just a POTS line from the local telco (Alltel).

 I've got to imagine it is some kind of momentary dip or spike in the line
 voltage that is coming through the FXO port.

Curious that it has recently (past couple months) started happening while
the SPA3k has been in service since April.

Curiuser and curiouser...

Paul
-- 
Paul Dugas, Computer Engineer   Dugas Enterprises, LLC
[EMAIL PROTECTED] phone: 404-932-1355   522 Black Canyon Park
http://dugas.cc fax: 866-751-6494   Canton, GA 30114 USA
--
Onsite at GDOT W.Annex 404-463-2860 x199
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[Asterisk-Users] Question about 3Com(r) 3101 Basic Phone

2005-10-03 Thread Jorge Cisneros

 Hi, i have one question, the 3Com 3101 Basic Phone work with
asterisk, if so i any a especial firmware o another thing. And wath
other 3com ip phone product work with asterisk. I think is a good idea
to create a list with the all voip device and the status with asterisk.

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Re: [Asterisk-Users] X100p Problem, randomly hungup pstn line

2005-10-03 Thread pcman theMan

This is my zaptel and zapata. In my logger.conf this is what is enabled :
full = notice,warning,error,debug,verbose.

How can you turn on the log in  chan_zap.c and where you can access
it. You can see i'm a newbee :-)

Thanks for your help

Pierre



;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]

language=en
context=from-pstn
signalling=fxs_ks
rxwink=300

; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
;callwaiting=yes
;usecallingpres=yes
;callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
busydetect=yes
busycount=20
callprogress=yes
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;Include AMP configs
#include zapata_additional.conf
hanguponpolarityswitch
;Include genzaptelconf configs
#include zapata-auto.conf
;channel=1
callerid=asreceived


# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Span 1: WCFXO/0 Generic Clone Board 1
fxsks=1

# Global data

loadzone= us
defaultzone = us
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[Asterisk-Users] asterisk behind Linux iptables with masquerading and forwarding on

2005-10-03 Thread Bartosz Wegrzyn - asterisk

Hi,

I have this setup

DSL ROUTERLINUX-ASTERISK

LINUX acts as a router with this config:
ppp0 - internet interface (public)
eth1 - private interface: 192.168.1.254

asterisk interface 192.168.1.251

settings on LINUX:
iptables -t nat -A POSTROUTING -o ppp0 -j MASQUERADE
echo 1  /proc/sys/net/ipv4/ip_forward
iptables -t nat -A PREROUTING -p udp --dport 5060 -i ppp0 -j DNAT --to
192.168.1.251
iptables -t nat -A PREROUTING -p udp --dport 1:2 -i ppp0 -j DNAT
--to 192.168.1.251

Before I had this setup I had the same config, but instead of LINUX i used
the Wr54G router with port forwarding on. It looks like that I
misconfigured the iptables, but I dont know what I did wrong. Do I have to
add extra translation settings.  Thanks

Any ideas???


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[Asterisk-Users] Nufone

2005-10-03 Thread Crystal Stream, Incorporated

After -- IAX2/NuFone/3 is making progress passing it
to SIP/3044-bcd0 I'm getting a Busy tone and it's
not even connecting the call.


-- Executing Macro(SIP/3044-bcd0,
outvoip-2|1800759) in new stack
-- Executing SetCIDName(SIP/3044-bcd0, X X X|a)
in new stack
-- Executing SetCIDNum(SIP/3044-bcd0,
8663xx3|a) in new stack
-- Executing Authenticate(SIP/3044-bcd0, xx)
in new stack
-- Playing 'agent-pass' (language 'en')
-- Playing 'auth-thankyou' (language 'en')
-- Executing Monitor(SIP/3044-bcd0, wav) in new
stack
-- Executing Ringing(SIP/3044-bcd0, ) in new stack
-- Executing Wait(SIP/3044-bcd0, 2) in new stack
-- Executing Dial(SIP/3044-bcd0,
IAX2/[EMAIL PROTECTED]/1800759) in new stack
-- Called [EMAIL PROTECTED]/1800759
-- Call accepted by 66.225.202.72 (format ulaw)
-- Format for call is ulaw
-- IAX2/NuFone/3 is making progress passing it to
SIP/3044-bcd0
-- Hungup 'IAX2/NuFone/3'
== Spawn extension (macro-outvoip-2, s, 7) exited
non-zero on 'SIP/3044-bcd0' in macro 'outvoip-2'
== Spawn extension (crystal-sip, 8800759, 1)
exited non-zero on 'SIP/3044-bcd0'

x*CLI iax2 show peers
Name/UsernameHost Mask
Port  Status
voicepulse2/Fbg  66.234.228.166  (S)  255.255.255.255 
4569  Unmonitored
voicepulse1/Fbg  66.234.228.160  (S)  255.255.255.255 
4569  Unmonitored
NuFone   66.225.202.72   (S)  255.255.255.255 
4569  Unmonitored





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Re: [Asterisk-Users] 911 Q

Think you might have jumped to a conclusion that might not be valid.
If the telco can handle a PRI and will accept callerid from you,
and each unit has a valid telephone number, then the telco can populate
the callerid database with names. Those are the only two items the
telco can provide in real time.

The 911 center manager can better tell you exactly how they populate
their database with street addresses and unit numbers. That process is
likely an external non-automated process, or, the local telco is giving
them the info via a electronic/paper copy of their service order. But,
he's the only one that can tell you exactly how that works for his
center.

So, don't give. Go to the 911 manager and do some research; then go to
his contact at the telco to get the real facts.


 Thank you - while not directly an answer to my question, it directly
 addresses the root of my question, pointing me where I'll need to go to
 dig deeper.  It also tells me what we didn't want to hear, that there's
 a very good possibility that we simply won't be able to ensure that the
 911 call center can tell which unit a call comes from without verbal
 specification from the caller.
 
 j
 
 On Sun, 2005-10-02 at 08:13 -0600, Rich Adamson wrote:
 
  Asterisk is more then capable of sending the appropriate callerid info
  to any remote site including 911 centers. However, there is a telco between
  asterisk and the 911 center that may not have realistic policies/systems
  to accept and forward that callerid. So, your objective becomes one of
  what the telco will allow you to do (and their support of your objective).
  
  As one example only, the telco might have a switch that does not have
  PRI capabilities (I know of many of these), but they provide ANI info
  to the 911 centers since that _might_ be the only data they can provide.
  If that were the case in your environment, it doesn't make any difference
  what you do with asterisk, it won't be supported.
  
  I know from practical experience that a telco's switch (in most cases)
  will accept calleridnum via a PRI, but on most central office switches
  its an option that needs to be turned on. (Local telco policy _might_
  say they will never do that.) Once that option is turned on, you can
  send almost anything to them in the form of calleridnum.
  
  The callerid name is a different story.  The central office switch that
  _terminates_ any call (including 911 calls) will have a mechanism to do
  a database lookup/dip, and if that database has not been populated with
  an appropriate callerid name, will not provide callerid names to the
  911 center (or anyone else). That essentially says you can program
  asterisk to send anything that you want from a callerid name perspective
  and it will be ignored in the US. In very general terms, only telco 
  personnal have the access to update the callerid database, and usually
  that is limited to the CO prefixes they support.
  
  Also keep in mind that not all 911 centers are the same from a technical
  perspective. They certainly accept callerid numbers, but they may have
  their own local database for names (etc), or, they may also do a database
  lookup from some distant database.  If you think about those customers
  that subscribe to callerid blocking, cell phones  gps, and the 
  requirements 
  of 911 centers, its not hard to visualize several different 911 
  implementation approaches.
  
  Talk to a knowledgable telco person (might be somewhat difficult to find
  the appropriate person), and talk to the 911 center manager to better
  understand what options you might have available. I'd start with the
  911 manager as he will know a telco person that understands the
  technical requirements.
  
  
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Re: [Asterisk-Users] Adding Cepstral to Asterisk

2005-10-03 Thread pbx

Then did you do a make clean / make / make install?

Then do show applications at the CLI prompt after you have restarted
asterisk.

service asterisk stop
service asterisk start

...

 I downloaded Cepstral to my Asterisk Box.  I did the install and let it
 install to /opt/swift.

 I brought down a new CVS-HEAD as of today 10/1.

 I added APPS+=app_cepstral.so into the Makefile in
 /usr/src/asterisk/apps/Makefile

 Like:

 # Obsolete things...
 #
 #APPS+=app_sql_postgres.so
 #APPS+=app_sql_odbc.so
 APPS+=app_cepstral.so
 #

 I did this piece but wasn't sure exactly what part of the Makefile I was
 to
 add it in so I added it in here:

 Towards the top of the file where it talks obsolete programs are commented
 out.
 And then after the section that compiles voicemail add:

 app_cepstral.so: app_cepstral.c
   $(CC) -D_GNU_SOURCE -shared -Xlinker -x -o $@ $ -lz -lm -lswift
 -lceplex_us -lceplang_en -lz -ldl -L/opt/swift/lib -I/opt/swift/include

 Make sure the $(CC) line starts with a tab, not spaces.


 I didn't see a lot about voicemail:

 app_sql_odbc.so: app_sql_odbc.o
 $(CC) $(SOLINK) -o $@ $ -lodbc

 app_cepstral.so: app_cepstral.c
 $(CC) -D_GNU_SOURCE -shared -Xlinker -x -o $@ $ -lz -lm -lswift
 -lceplex_us -lceplang_en -lz -ldl -L/opt/swift/lib -I/opt/swift/include

 look:   look.c
 $(CC) -pipe -O6 -g look.c -o look -lncurses


 I checked the /etc/ld.so.conf file for a line like: opt/swift/lib in the
 file.  It wasn't there so I added it:

 include ld.so.conf.d/*.conf
 /opt/swift/lib


 I ran ldconfig when I was done.

 I can't see that Cepstral was added into Asterisk and I was wondering what
 I
 have done wrong that it doesn't work.

 Thanks.









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R: [Asterisk-Users] Diva

2005-10-03 Thread Giordano Grandis









Which models of Diva
could work with CAPI and asterisk?



Thanks





Giordano











Da:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di [EMAIL PROTECTED]
Inviato: sabato 1 ottobre 2005
23.46
A: asterisk-users@lists.digium.com
Oggetto: RE: [Asterisk-Users] Diva





Nope. At least I tried and never could get
it working. It's a semiactive.









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis
Sent: Friday, September 30, 2005
6:59 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Diva

Hi all,

just a question: can i use this kind of
diva for asterisk?



00:14.0 Network controller: Eicon Networks
Corporation Diva ISDN Pro 3.0 PCI



Thanks all



Giordano 








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Re: [Asterisk-Users] Revieving some fax problems

2005-10-03 Thread Alexandre Leclerc

I would say the problem here could fall in this category.

Jason Walker a écrit :
 I have run into a similar situation. One of our older faxes at the office
 seems to not work with spandsp module. The newer faxes work just fine. 
 
 When I watch the logs, there appears to be communication from * requesting
 the fax to slow down. When the fax machine does not respond, * seems to
 say forget it and fail on the retrieval.
 
 I have not come up with a fix...regardless of rx/tx gains on the zaptel
 cards.

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Re: [Asterisk-Users] Adding Cepstral to Asterisk

2005-10-03 Thread Wojciech Tryc


I am not following...
Why would you need to integrate Cepstral directly into Asterisk? Just to be 
able to call it as Asterisk app from your dialplan? I am running Cepstral 
and calling it through the System call.

Thanks,
Wojtek
- Original Message - 
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Monday, October 03, 2005 11:27 AM
Subject: Re: [Asterisk-Users] Adding Cepstral to Asterisk


Then did you do a make clean / make / make install?

Then do show applications at the CLI prompt after you have restarted
asterisk.

service asterisk stop
service asterisk start

...


I downloaded Cepstral to my Asterisk Box.  I did the install and let it
install to /opt/swift.

I brought down a new CVS-HEAD as of today 10/1.

I added APPS+=app_cepstral.so into the Makefile in
/usr/src/asterisk/apps/Makefile

Like:

# Obsolete things...
#
#APPS+=app_sql_postgres.so
#APPS+=app_sql_odbc.so
APPS+=app_cepstral.so
#

I did this piece but wasn't sure exactly what part of the Makefile I was
to
add it in so I added it in here:

Towards the top of the file where it talks obsolete programs are commented
out.
And then after the section that compiles voicemail add:

app_cepstral.so: app_cepstral.c
$(CC) -D_GNU_SOURCE -shared -Xlinker -x -o $@ $ -lz -lm -lswift
-lceplex_us -lceplang_en -lz -ldl -L/opt/swift/lib -I/opt/swift/include

Make sure the $(CC) line starts with a tab, not spaces.


I didn't see a lot about voicemail:

app_sql_odbc.so: app_sql_odbc.o
$(CC) $(SOLINK) -o $@ $ -lodbc

app_cepstral.so: app_cepstral.c
$(CC) -D_GNU_SOURCE -shared -Xlinker -x -o $@ $ -lz -lm -lswift
-lceplex_us -lceplang_en -lz -ldl -L/opt/swift/lib -I/opt/swift/include

look:   look.c
$(CC) -pipe -O6 -g look.c -o look -lncurses


I checked the /etc/ld.so.conf file for a line like: opt/swift/lib in the
file.  It wasn't there so I added it:

include ld.so.conf.d/*.conf
/opt/swift/lib


I ran ldconfig when I was done.

I can't see that Cepstral was added into Asterisk and I was wondering what
I
have done wrong that it doesn't work.

Thanks.









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[Asterisk-Users] no audio on fxo line

2005-10-03 Thread phpmechanic


Hi,

I got back from two weeks away and appear to have lost audio on my 
tdm411 fxo. Everything was working properly when I left. I checked the 
logs, config files and can't see any problems, the zap channels and 
modules are all loaded properly, asterisk starts without probs and 
everything looks sweet on the colsole with -c when I make calls, 
but I just don't hear a dialtone or any audio anymore.  I tried opening 
the sound monitor and that looks as though appropriate sounds are being 
sent.  When I pick up the handset all I can hear is a slight crackling 
noise. The fxo line rings but I don't hear audio through any phones 
connected to it..


I'm not really sure what I can try to resolve this - has anybody got 
some suggetion?

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Re: [Asterisk-Users] Re:Any way to not overwrite sound files on compile?

2005-10-03 Thread Dave Cotton

On Sat, 2005-10-01 at 07:39 -0600, Rich Adamson wrote:
   

 
 I believe you meant to say make update. upgrade is not a defined
 parameter.

No, I meant to say exactly what I said.

Read the F Makefile :), line 677
 
upgrade: all bininstall
 

-- 
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Re: [Asterisk-Users] updating display of a hardphone based on agentslogging in

2005-10-03 Thread Mark Elkins

I'm also using SNOM320/360 phones. Ideally - set up one button to toggle
the Agent Status (in/out == On/Off) ???
Kinda make sense if app_devstate (or similar) made it into mainstrean
Asterisk - so line indication lamps could be used at will.

The SNOM320 is so ideal for Call Centres (the Headset control it gives
one) - I'm surprised that there is not a dedicated Agent Has Logged-in
icon... :-)

On Fri, 2005-08-26 at 10:20 +0200, Nils Ohlmeier wrote:
 On the Snom phones you can use a SIP MESSAGE to overwrite the idle screen 
 text 
 with a given text message. Maybe that is helpfull for your scenario.
 
 Regards
   Nils Ohlmeier

Nils (or anyone else) - how does one do this from Asterisk?

   You've got the Snom 320's, so maybe the most straight forward thing
  to do would be to use the Hint application with them to light a status
  LED when an agent is logged in and have it go dark when the agent is
  logged out.
   We are settng up a fair sized call center on Asterisk, but we are 
   having

-- 
  .  . ___. .__  Posix Systems - Sth Africa.  e.164 VOIP ready
 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

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Re: R: [Asterisk-Users] Diva

2005-10-03 Thread Armin Schindler

On Mon, 3 Oct 2005, Giordano Grandis wrote:
 Which models of Diva could work with CAPI and asterisk?

 - 'Diva Server' PCI cards with 'divas' driver from melware.net or Eicon source 
RPM
 - passive Diva cards supported by mISDN

Armin
 
 Thanks
 
  
 
 Giordano
 
 
 
 Da: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Per conto di
 [EMAIL PROTECTED]
 Inviato: sabato 1 ottobre 2005 23.46
 A: asterisk-users@lists.digium.com
 Oggetto: RE: [Asterisk-Users] Diva
 
  
 
 Nope. At least I tried and never could get it working.  It's a
 semiactive.
 
  
 
 
 
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Giordano
 Grandis
 Sent: Friday, September 30, 2005 6:59 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Diva
 
 Hi all,
 
 just a question:   can i use this kind of diva for asterisk?
 
  
 
 00:14.0 Network controller: Eicon Networks Corporation Diva ISDN Pro 3.0
 PCI
 
  
 
 Thanks all
 
  
 
 Giordano 
 
  
 
 
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Re: [Asterisk-Users] SPA-3000 generating one-ring calls


 This is a wierd one.  Can't figure it out.  I have an SPA-3000 at the
 house handling my incoming line.  It's setup to direct the incoming call
 to asterisk.  Works great 99% of the time.
 
 A few times a day, I'm getting calls that ring once internally and are
 then hungup.  I managed to get a detailed log [1] of what's happening
 today and it looks to me that the SPA is acting wierd.  Can someone verify
 this for me?
 
 I looks to me that the Sipura is just CANCEL'ing the call shortly (2 secs
 in this example) after setting it up.  I'm looking for someone to verify
 this before I stop looking at Asterisk as the cause and focus on the SPA.
 
Not likely anyone is going to comment on this without looking at your
traces, s/w versions, config detail, etc.  There are just too many ways
to configure an spa and guessing at what you've done is impossible.

FWIW, mine and others are working fine.


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Re: [Asterisk-Users] Nufone

2005-10-03 Thread Tom Vile

how many digits is your callerid passing to the trunk? I am seeing 11
8663xx3 is that correct? I had an issue last week with
passing to many digits to my provider and the call would hang up
immediately. 

You could also turn debugging on for this so we can get a better log.

iax2 debug peer nufoneOn 10/3/05, Crystal Stream, Incorporated [EMAIL PROTECTED] wrote:
After -- IAX2/NuFone/3 is making progress passing itto SIP/3044-bcd0 I'm getting a Busy tone and it's
not even connecting the call.-- Executing Macro(SIP/3044-bcd0,outvoip-2|1800759) in new stack-- Executing SetCIDName(SIP/3044-bcd0, X X X|a)
in new stack-- Executing SetCIDNum(SIP/3044-bcd0,8663xx3|a) in new stack-- Executing Authenticate(SIP/3044-bcd0, xx)in new stack-- Playing 'agent-pass' (language 'en')
-- Playing 'auth-thankyou' (language 'en')-- Executing Monitor(SIP/3044-bcd0, wav) in newstack-- Executing Ringing(SIP/3044-bcd0, ) in new stack-- Executing Wait(SIP/3044-bcd0, 2) in new stack
-- Executing Dial(SIP/3044-bcd0,IAX2/[EMAIL PROTECTED]/1800759) in new stack-- Called [EMAIL PROTECTED]/1800759-- Call accepted by 66.225.202.72
 (format ulaw)-- Format for call is ulaw-- IAX2/NuFone/3 is making progress passing it toSIP/3044-bcd0-- Hungup 'IAX2/NuFone/3'== Spawn extension (macro-outvoip-2, s, 7) exitednon-zero on 'SIP/3044-bcd0' in macro 'outvoip-2'
== Spawn extension (crystal-sip, 8800759, 1)exited non-zero on 'SIP/3044-bcd0'x*CLI iax2 show peersName/UsernameHost
MaskPortStatusvoicepulse2/Fbg66.234.228.166(S)255.255.255.2554569Unmonitoredvoicepulse1/Fbg
66.234.228.160(S)255.255.255.2554569UnmonitoredNuFone 66.225.202.72 (S)255.255.255.255
4569Unmonitored__Yahoo! Mail - PC Magazine Editors' Choice 2005http://mail.yahoo.com___
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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Fax: 518-631-2856
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Re: [Asterisk-Users] Asterisk for Man-In-The-Middle Trunk Side Call Recording?

2005-10-03 Thread Dinesh Nair




On 09/30/05 03:12 Verlin Henderson said the following:

Xeon server (most likely a Dell PowerEdge 2800, 2850, or similar) with a
large amount of RAM and RAID-1 SCSI setup. We would add three TE411P or
TE410P cards and implement something similar to Matt Roth's setup, but on a
smaller scale.


has the limit on 254 zap channels per server been removed ? admittedly, i 
may have missed the change where this occured, but it's certainly there in 
zaptel from a few months back.


--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
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Re: [Asterisk-Users] SPA-3000 generating one-ring calls

That's interesting for sure. I'd bet if you had some way to monitor what was going on with the FXO (voltage) side of things you'd probably find that something is happening that is causing the spa3k to believe that it's receiving ring voltage on the line. You can tune these settings in International Control in the Advanced/Admin and PSTN Line tab section of the Spa3k config, but you've got to know what and for how long you're receiving something first before you know what to tune. 

On 10/3/05, Paul Dugas [EMAIL PROTECTED] wrote:
On Mon, October 3, 2005 10:10 am, BJ Weschke wrote: What is the FXO port connected to? An ATA from another VoIP provider?
It's just a POTS line from the local telco (Alltel). I've got to imagine it is some kind of momentary dip or spike in the line voltage that is coming through the FXO port.Curious that it has recently (past couple months) started happening while
the SPA3k has been in service since April.Curiuser and curiouser...Paul--Paul Dugas, Computer Engineer Dugas Enterprises, LLC[EMAIL PROTECTED] phone: 404-932-1355 522 Black Canyon Park
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Re: [Asterisk-Users] Fritz, mISDN, Help

2005-10-03 Thread Dave Cotton

On Fri, 2005-09-30 at 09:38 +0100, Derek Conniffe wrote:

 
 Is anyone out there running two AVM Fritz ISDN cards?  

Yes

 Are you using a 2.6.XX kernel?  

No

 How are you doing it?

Easily :)

Really just carefully follow the instructions in the hack you've
already mentioned.

It works, but I did chicken out on 3.


-- 
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Re: [Asterisk-Users] no audio on fxo line


 I got back from two weeks away and appear to have lost audio on my 
 tdm411 fxo. Everything was working properly when I left. I checked the 
 logs, config files and can't see any problems, the zap channels and 
 modules are all loaded properly, asterisk starts without probs and 
 everything looks sweet on the colsole with -c when I make calls, 
 but I just don't hear a dialtone or any audio anymore.  I tried opening 
 the sound monitor and that looks as though appropriate sounds are being 
 sent.  When I pick up the handset all I can hear is a slight crackling 
 noise. The fxo line rings but I don't hear audio through any phones 
 connected to it..
 
 I'm not really sure what I can try to resolve this - has anybody got 
 some suggetion?

Check to see what revision of TDM card you have. The rev E/F cards and
earlier had a problem that sounds just like what you are hearing and
the only fix is to replace the card. Contact support at digium.

The problem usually happens anywhere from one to two weeks after a
reboot. Stopping asterisk, unload and reload the drivers, and start
asterisk also clears the problem without a reboot.

Since the card has a two year warranty and hasn't been out that long,
you should be able to get digium to replace it at no cost.


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[Asterisk-Users] SIP-CPE Gateway

2005-10-03 Thread Bill Michaelson

Has anyone used the GSM-SIP gateway product produced by a company at 
sipcpe.com?  Any comments?



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Re: [Asterisk-Users] What does the error stale nonce' mean?

2005-10-03 Thread Bob Goddard

On Monday 03 Oct 2005 08:51, Olle E. Johansson wrote:
 Paul Conn wrote:
  I’m receiving the following error over and over, adnauseam:
 
 
 
  Oct  1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce
  received from ‘CNAME-CID sip:[EMAIL PROTECTED]’
 
 
 
  Does anyone know what “stale nonce” is?

 I've answered this question many times, so you should be able to find
 the answer...

 A stale nonce is when a device tries to re-authenticate with a nonce
 that is no longer valid. We are telling them that the nonce they used is
 invalid, and re-issue a new challenge and a fresh nonce. It's just an
 informative message, that I propably should move away to a debug level
 of some kind.

I wish someone had read a British dictionary before they
decided to use this word It make no sense at all.


B
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Re: [Asterisk-Users] Asterisk for Man-In-The-Middle Trunk Side Call Recording?

Yes. It's gone.
On 10/3/05, Dinesh Nair [EMAIL PROTECTED] wrote:
On 09/30/05 03:12 Verlin Henderson said the following: Xeon server (most likely a Dell PowerEdge 2800, 2850, or similar) with a
 large amount of RAM and RAID-1 SCSI setup. We would add three TE411P or TE410P cards and implement something similar to Matt Roth's setup, but on a smaller scale.has the limit on 254 zap channels per server been removed ? admittedly, i
may have missed the change where this occured, but it's certainly there inzaptel from a few months back.--Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/+==oOO--(_)--OOo==+| for a in past present future; do|
| for b in clients employers associates relatives neighbours pets; do || echo The opinions here in no way reflect the opinions of my $a $b.|| done; done|
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Re: [Asterisk-Users] SPA-3000 generating one-ring calls

2005-10-03 Thread Landon Stewart | Superb Internet Corp.

On Mon, October 3, 2005 12:44 pm, Rich Adamson wrote:
 Not likely anyone is going to comment on this without looking at your
 traces, s/w versions, config detail, etc.  There are just too many ways
 to configure an spa and guessing at what you've done is impossible.

Good point.  The trace of what happened is in available for donwload [1]
if anyone is curious.  I'm running Asterisk CVS HEAD last updated Sep-23. 
The SPA3k is running 3.1.5(GWb) firmware and is very close to the stock
config; have set NTP host, timezone, DST rule, Line 1
proxy/userID/password, PSTN Line proxy/userID/password and the
PSTN-to-VOIP Gateway settings.

 FWIW, mine and others are working fine.

This one used to work fine too :)

Thanks,

Paul

[1] http://dugas.cc/~pdugas/spa3k.log

-- 
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[EMAIL PROTECTED] phone: 404-932-1355   522 Black Canyon Park
http://dugas.cc fax: 866-751-6494   Canton, GA 30114 USA
--
Onsite at GDOT W.Annex 404-463-2860 x199
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[Asterisk-Users] suse 9.3 pro asterisk install from source problem

2005-10-03 Thread ashley wright









Hi,

Can any one help Im trying to install asterisk on
suse 9.3 pro from cvs release v1_0 version 1.0.9 and when I try to make
from the asterisk directory I get the following error.



Is there anybody that could give me a pointer as to what the
issue may be?







DDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\
-DBUSYDETECT_MARTIN
-DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC -c -o res_indications.o
res_indications.c

gcc -shared -Xlinker -x -o res_indications.so
res_indications.o

gcc -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include
-D_REENTRANT -D_GNU_SOURCE -O6 -march=i686
-DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-v1-0-09/30/05-22:31:05\
-DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\
-DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\
-DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\
-DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\
-DBUSYDETECT_MARTIN
-DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC -c -o res_monitor.o
res_monitor.c

gcc -shared -Xlinker -x -o res_monitor.so res_monitor.o

gcc -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include
-D_REENTRANT -D_GNU_SOURCE -O6 -march=i686
-DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-v1-0-09/30/05-22:31:05\
-DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\
-DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\
-DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\
-DBUSYDETECT_MARTIN -DZAPATA_MOH
-DOPENSSL_NO_KRB5 -fPIC -c -o res_agi.o res_agi.c

gcc -shared -Xlinker -x -o res_agi.so res_agi.o

make[1]: Leaving directory
`/usr/local/src/development/asteriskv1-0_1.0.9/asterisk/res'

make[1]: Entering directory
`/usr/local/src/development/asteriskv1-0_1.0.9/asterisk/channels'

gcc -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include
-D_REENTRANT -D_GNU_SOURCE -O6 -march=i686
-DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-09/30/05-22:31:05\
-DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\
-DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\
-DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\
-DBUSYDETECT_MARTIN -Wno-missing-prototypes
-Wno-missing-declarations -DZAPATA_PRI
-DIAX_TRUNKING -DCRYPTO -fPIC -c -o chan_modem.o
chan_modem.c

gcc -shared -Xlinker -x -o chan_modem.so chan_modem.o

gcc -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include
-D_REENTRANT -D_GNU_SOURCE -O6 -march=i686
-DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-09/30/05-22:31:05\
-DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\
-DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\
-DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\
-DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\
-DBUSYDETECT_MARTIN -Wno-missing-prototypes
-Wno-missing-declarations -DZAPATA_PRI
-DIAX_TRUNKING -DCRYPTO -fPIC -c -o chan_sip.o
chan_sip.c

chan_sip.c:9319: internal compiler error: output_operand:
invalid _expression_ as operand

Please submit a full bug report,

with preprocessed source if appropriate.

See URL:http://www.suse.de/feedback for
instructions.

{standard input}: Assembler messages:

{standard input}:123824: Warning: partial line at end of
file ignored

Preprocessed source stored into /tmp/ccxI4zbE.out file,
please attach this to your bugreport.

make[1]: *** [chan_sip.o] Error 1

make[1]: Leaving directory
`/usr/local/src/development/asteriskv1-0_1.0.9/asterisk/channels'

make: *** [subdirs] Error 1







Ashley Wright

Systems Engineer



[EMAIL PROTECTED]

www.OciusB2.com



OciusB2 Limited

The Heath

Runcorm

Cheshire

WA7 4QX



Tel:0870 7578700

Fax: 01928 515401



Note.
This email is confidential, may be legally privileged, and is for the intended
recipient only. Access, disclosure, copying, distribution, or reliance on any
of it by anyone else is prohibited and may be a criminal offence. Please delete
if obtained in error and email confirmation to the

sender.








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[Asterisk-Users] Inter Asterisk IAX2

2005-10-03 Thread Geo

Hello,

Would like to use IAX /IAX2 to transport 30 channels inter Asterisk.
I don't have any experience with that, so can someone help ??   
How much bw do I need and what latency for SIP G711 to IAX and vice-versa , ... 
etc  ?
Thanks in advance for any info,

Geo







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[Asterisk-Users] TDMoE help with Alarms...

2005-10-03 Thread pbx

I have configured TDMoE sucessfully and I am able to make a Zap connection
from one box to the other.

The question I have is..

I'm getting repeated errors every second on both systems..

Oct  3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected
alarm on channel 1: No Alarm
Oct  3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected
alarm on channel 2: No Alarm
Oct  3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected
alarm on channel 3: No Alarm
Oct  3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected
alarm on channel 4: No Alarm
Oct  3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected
alarm on channel 5: No Alarm
Oct  3 09:53:16 NOTICE[4409]: chan_zap.c:6247 handle_init_event: Alarm
cleared on channel 1
Oct  3 09:53:16 NOTICE[4409]: chan_zap.c:6247 handle_init_event: Alarm
cleared on channel 2
Oct  3 09:53:16 NOTICE[4409]: chan_zap.c:6247 handle_init_event: Alarm
cleared on channel 3
Oct  3 09:53:16 NOTICE[4409]: chan_zap.c:6247 handle_init_event: Alarm
cleared on channel 4
Oct  3 09:53:16 NOTICE[4409]: chan_zap.c:6247 handle_init_event: Alarm
cleared on channel 5


What is causing these errors?

When i do a zttool it shows that there are no errors...

Thanks...




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Re: [Asterisk-Users] 911 Q

2005-10-03 Thread Andrew Kohlsmith

On Monday 03 October 2005 12:17, Rich Adamson wrote:
 Think you might have jumped to a conclusion that might not be valid.
 If the telco can handle a PRI and will accept callerid from you,
 and each unit has a valid telephone number, then the telco can populate
 the callerid database with names. Those are the only two items the
 telco can provide in real time.

I have some information from the 911 service manager for Bell Canada in 
Eastern Ontario.  

Basically the Public Service Automatic Location Indentification database 
(PSALI) only has allocations for BTNs (Billing Telephone Numbers) -- there 
are no ALI entries for DIDs from Bell Canada at this time, and there is no 
plan to do this.   Basically if you set your outgoing ANI to a DID the PSAP 
office will have no address information, and indeed the switch may end up 
overwriting your ANI with the BTN.

Since DIDs do not have an address associated with them  (makes sense, they are 
only inward-numbers by design), you can convert DIDs to LDNs (Local 
Directory Number, same thing but has a directory (address) associated with 
it) -- the unfortunate side-effect of that is that LDNs are all billed 
separately so you would receive a separate bill for every LDN on a PRI.

There is a service (of course!) being offered where you can provide 
specifically-formatted records for the PSALI database.  It's not cheap, it's  
a $2000 setup fee and (IIRC) $500/mo for up to 500 record changes, and a 
two-year contract minimum.  (These figures might be off, it's from memory.)  
However if you subscribe to this service you can assign any municipal address 
to any number and it will make its way into the PSALI database, which is what 
all the primary PSAP offices use to get the address information before 
routing the call to the appropriate secondary PSAP office.

At least with Bell Canada, this is the only way to get your user's address 
information into the database used by the primary PSAP offices.  The 
alternative, of course, is to set up your own primary PSAP system and then 
you can use whatever database and organization system you want, and redirect 
calls to the appropriate secondary PSAP office yourself.

-A.
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Re: [Asterisk-Users] updating display of a hardphone based on agentslogging in

2005-10-03 Thread Philipp von Klitzing

Hi!

On Mon, Oct 03, 2005 at 05:41:38PM +0200, Mark Elkins wrote:
 I'm also using SNOM320/360 phones. Ideally - set up one button to toggle
 the Agent Status (in/out == On/Off) ???
 Kinda make sense if app_devstate (or similar) made it into mainstrean
 Asterisk - so line indication lamps could be used at will.

Just another thought: You could also simply use the snom DND button,
define an Action URL on the phone for both DND on and off and let that
web page (.cgi, .php or whatever you prefer) sign your agent on and off.

Cheers, Philipp

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Re: [Asterisk-Users] Nufone

2005-10-03 Thread Crystal Stream, Incorporated

crystalstream*CLI
-- Executing Macro(SIP/3044-5300,
outvoip-2|1800759) in new stack
-- Executing SetCIDName(SIP/3044-5300, CRYSTAL
STREAM NET|a) in new st ack
-- Executing SetCIDNum(SIP/3044-5300,
866xxx|a) in new stack
-- Executing Authenticate(SIP/3044-5300,
123987) in new stack
-- Playing 'agent-pass' (language 'en')
-- Playing 'auth-thankyou' (language 'en')
-- Executing Monitor(SIP/3044-5300, wav) in
new stack
-- Executing Ringing(SIP/3044-5300, ) in new
stack
-- Executing Wait(SIP/3044-5300, 2) in new
stack
-- Executing Dial(SIP/3044-5300,
IAX2/[EMAIL PROTECTED]/1800759 ) in new
stack
-- Called [EMAIL PROTECTED]/1800759
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type:
IAX Subclass: NEW
   Timestamp: 3ms  SCall: 1  DCall: 0
[66.225.202.72:4569]
   VERSION : 2
   CALLED NUMBER   : 1800759
   CALLING NUMBER  : 8663113060
   LANGUAGE: en
   USERNAME: username-hidden
   FORMAT  : 4
   CAPABILITY  : 63502
   ADSICPE : 2
   DATE TIME   : 188966086

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type:
IAX Subclass: AUTHREQ
   Timestamp: 3ms  SCall: 00061  DCall: 1
[66.225.202.72:4569]
   AUTHMETHODS : 2
   CHALLENGE   : 150617580
   USERNAME: username-hidden

Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type:
IAX Subclass: AUTHREP
   Timestamp: 00033ms  SCall: 1  DCall: 00061
[66.225.202.72:4569]
   MD5 RESULT  : c8214533976d4dec8b233543dac0eaac

Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type:
IAX Subclass: ACCEPT
   Timestamp: 00036ms  SCall: 00061  DCall: 1
[66.225.202.72:4569]
   FORMAT  : 4

-- Call accepted by 66.225.202.72 (format ulaw)
-- Format for call is ulaw
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type:
IAX Subclass: ACK
   Timestamp: 00036ms  SCall: 1  DCall: 00061
[66.225.202.72:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type:
VOICE   Subclass: 4
   Timestamp: 00060ms  SCall: 1  DCall: 00061
[66.225.202.72:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type:
VOICE   Subclass: 4
   Timestamp: 00080ms  SCall: 00061  DCall: 1
[66.225.202.72:4569]
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type:
IAX Subclass: ACK
   Timestamp: 00080ms  SCall: 1  DCall: 00061
[66.225.202.72:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type:
IAX Subclass: ACK
   Timestamp: 00060ms  SCall: 00061  DCall: 1
[66.225.202.72:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type:
CONTROL Subclass: (15?)
   Timestamp: 00123ms  SCall: 00061  DCall: 1
[66.225.202.72:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type:
IAX Subclass: ACK
   Timestamp: 00123ms  SCall: 1  DCall: 00061
[66.225.202.72:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 003 Type:
CONTROL Subclass: (14?)
   Timestamp: 01423ms  SCall: 00061  DCall: 1
[66.225.202.72:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type:
IAX Subclass: ACK
   Timestamp: 01423ms  SCall: 1  DCall: 00061
[66.225.202.72:4569]
-- IAX2/NuFone/1 is making progress passing it to
SIP/3044-5300
Oct  3 12:38:19 WARNING[49584]: app_dial.c:372
wait_for_answer: Unable to forwar d frame
-- Hungup 'IAX2/NuFone/1'
  == Spawn extension (macro-outvoip-2, s, 7) exited
non-zero on 'SIP/3044-5300' in macro 'outvoip-2'
  == Spawn extension (crystal-sip, 8800759, 1)
exited non-zero on 'SIP/3044- 5300'


--- Tom Vile [EMAIL PROTECTED] wrote:

 how many digits is your callerid passing to the
 trunk? I am seeing 11
 8663xx3 is that correct? I had an issue last
 week with passing to many
 digits to my provider and the call would hang up
 immediately.
 
 You could also turn debugging on for this so we can
 get a better log.
 
 iax2 debug peer nufone
 
 On 10/3/05, Crystal Stream, Incorporated
 [EMAIL PROTECTED] wrote:
 
  After -- IAX2/NuFone/3 is making progress passing
 it
  to SIP/3044-bcd0 I'm getting a Busy tone and
 it's
  not even connecting the call.
 
  
  -- Executing Macro(SIP/3044-bcd0,
  outvoip-2|1800759) in new stack
  -- Executing SetCIDName(SIP/3044-bcd0, X X
 X|a)
  in new stack
  -- Executing SetCIDNum(SIP/3044-bcd0,
  8663xx3|a) in new stack
  -- Executing Authenticate(SIP/3044-bcd0,
 xx)
  in new stack
  -- Playing 'agent-pass' (language 'en')
  -- Playing 'auth-thankyou' (language 'en')
  -- Executing Monitor(SIP/3044-bcd0, wav) in
 new
  stack
  -- Executing Ringing(SIP/3044-bcd0, ) in new
 stack
  -- Executing Wait(SIP/3044-bcd0, 2) in new
 stack
  -- Executing Dial(SIP/3044-bcd0,
  IAX2/[EMAIL PROTECTED]/1800759) in new stack
  -- Called [EMAIL PROTECTED]/1800759
  -- Call accepted by 66.225.202.72
 http://66.225.202.72 (format ulaw)
  -- Format for call is ulaw
  -- IAX2/NuFone/3 is making progress passing it to
  SIP/3044-bcd0
  -- Hungup 'IAX2/NuFone/3'
  == Spawn extension (macro-outvoip-2, s, 7) exited
  non-zero on 'SIP/3044-bcd0'

[Asterisk-Users] Which hardware configuration? How would this work?

Hello Everyone,

Please accept my appologies - I've been reading through the handbook
and the online documentation / mailing list archives and can't quite
get my own answer to these inquiries... The biggest mystery is
how the existing handsets are connected to a new machine running
Asterisk.

Background:
- The phone system we have is horribly out of date and may pack-it-in any day now.
- Existing PBX system (AltiReach running on NT4) but we plan on replacing this server entirely and ditching the old PCI cards but keeping the hand sets (approximately 30 Nortel hand sets).

- We have 12 regular phone lines coming into this system
- We have satelite offices that could be VOIP after the system is implemented.

What is the best hardware configuration for this? Should we get a
T1? Which cards/hardware should we use? We are currently unclear
on how the hand sets connect to the system but moderately clear on how
the phone lines would connect to the box. Some information
sources or direct examples of how to switch from a 30 handset office to
an Asterisk system would be awesome. Once we replace our current
setup we will delve into the extended features/options available.
VOIP is probably the most important one after we switch systems
entirely.

If there is anything else I can provide to help you help me I will reply as soon as possible.
-- Landon StewartSuperb Internet Corporation
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[Asterisk-Users] SIP qualify question.

2005-10-03 Thread Ray Van Dolson

When qualify is set to yes in sip.conf for a friend and the OPTIONS packet
gets returned with an ICMP port unreachable message, what is the behavior of
Asterisk?

It looks to me like Asterisk tries sending the OPTION request again right away
(well within a second or two).

Some of our devices are being Linux firewalls that make use of iptables to do
portforwarding.  This generates entries in /proc/net/ip_contrack.  From time
to time, these entries get out of whack or a connection gets stale and inbound
requests start getting rejected with port unreachable messages.

In order to get things working again, I need to expire the corresponding entry
in ip_conntrack (I have my timeouts set low -- to 15 seconds).  The problem is
is that Asterisk is sendint the qualify requests so quickly that the timer
keeps resetting on my ip_conntrack entry and never expires.  I have to either
reboot the Linux device to clear the entry manually or block requests from
Asterisk for 15 seconds.  Then things work fine again.

Wondering if there's a good way to make Asterisk back off a little if a SIP
OPTIONS request as part of qualify doesn't get through.  I suppose modifying
the source code is our best bet.

Setting qualify to a milliseconds value doesn't appear to affect these
retransmits btw.

Thanks!
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[Asterisk-Users] Hangup not detected on callback

2005-10-03 Thread asterisk


Hi,

I'm trying to set up a call-back system using auto-dialout files. I 
want the call to be terminated when a specific timeout (defined in the 
.call file) is detected. Both parties should then be hangup.

The problem is that the timeout is never detected... How to solve this?

Thank you,

Pierre

.call file
--

Channel: IAX2/:@xxx.xxx.xxx.xxx/01
Callerid: 1
MaxRetries: 5
RetryTime: 60
WaitTime: 30
Context: test
Extension: 02
Priority: 1
SetVar: ato=30
SetVar: act=testaccount

extensions.conf
---

[test]
exten = _XX,1,SetAccount(${act})
exten = _XX,2,AbsoluteTimeout(${ato})
exten = _XX,3,Answer()
exten = _XX,4,Dial(IAX2/:@xxx.xxx.xxx.xxx/${EXTEN})
exten = _XX,5,Congestion()
exten = _XX,102,Busy()

exten = s,1,DigitTimeout,10
exten = s,2,ResponseTimeout,10

exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
exten = T,1,Playback(vm-goodbye)
exten = T,2,Hangup

CLI output
--

-- Attempting call on IAX2/:@xxx.xxx.xxx.xxx/01 for 
[EMAIL PROTECTED]:1 (Retry 1)

-- Call accepted by xxx.xxx.xxx.xxx (format ulaw)
-- Format for call is ulaw
Channel IAX2/xxx.xxx.xxx.xxx:4569/1 was answered.
-- Executing SetAccount(IAX2/xxx.xxx.xxx.xxx:4569/1, 
testaccount) in new stack
-- Executing AbsoluteTimeout(IAX2/xxx.xxx.xxx.xxx:4569/1, 30) 
in new stack

-- Set Absolute Timeout to 30
-- Executing Answer(IAX2/xxx.xxx.xxx.xxx:4569/1, ) in new stack
-- Executing Dial(IAX2/xxx.xxx.xxx.xxx:4569/1, 
IAX2/:@xxx.xxx.xxx.xxx/02) in new stack

-- Called :@xxx.xxx.xxx.xxx/02
-- Call accepted by xxx.xxx.xxx.xxx (format ulaw)
-- Format for call is ulaw
-- IAX2/xxx.xxx.xxx.xxx:4569/2 is ringing
-- IAX2/xxx.xxx.xxx.xxx:4569/2 stopped sounds
-- IAX2/xxx.xxx.xxx.xxx:4569/2 answered IAX2/xxx.xxx.xxx.xxx:4569/1
-- Attempting native bridge of IAX2/xxx.xxx.xxx.xxx:4569/1 and 
IAX2/xxx.xxx.xxx.xxx:4569/2

-- Channel 'IAX2/xxx.xxx.xxx.xxx:4569/2' ready to transfer
-- Channel 'IAX2/xxx.xxx.xxx.xxx:4569/1' ready to transfer
-- Releasing IAX2/xxx.xxx.xxx.xxx:4569/1 and 
IAX2/xxx.xxx.xxx.xxx:4569/2

-- Hungup 'IAX2/xxx.xxx.xxx.xxx:4569/2'
  == Spawn extension (test, 02, 4) exited non-zero on 
'IAX2/xxx.xxx.xxx.xxx:4569/1'
Oct  3 19:14:04 NOTICE[1041]: chan_iax2.c:1378 iax2_destroy: Avoiding 
IAX destroy deadlock

-- Hungup 'IAX2/xxx.xxx.xxx.xxx:4569/1'
Oct  3 19:14:04 NOTICE[1092]: pbx_spool.c:242 attempt_thread: Call 
completed to IAX2/:@xxx.xxx.xxx.xxx/01


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[Asterisk-Users] Need help with Cisco 7960

2005-10-03 Thread Christian


Hi all,
Does anyone know if it is possible to disable the pound key on the 7960 to 
not place calls so that other services can be used in Asterisk, such as call 
forwarding. Any info is apreciated, many thanks! 


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Re: [Asterisk-Users] SPA-841 Decode Latency?

2005-10-03 Thread alan

 Subject: Re: [Asterisk-Users] SPA-841 Decode Latency?

Luki [EMAIL PROTECTED] wrote:

  Does anyone have any familiarity with decode latency, specifically
  with Sipura devices? Why would it take 200+ms to decode a 20ms RTP
  packet? G.711u has existed for over 30 years, how hard could it be?

 Although I have never seem the decode latency to go above 30 ms on a
 LAN, it does go up to 80 ms if the Sipura device (phone or ATA) is
 connected via an Internet link which has jitter. So I don't know for
 sure, but my understanding is that it's the delay from the arrival of
 the packet until it's played; this is not due to the actual decoding
 but probably mostly due to the jitter buffer in the device, which is
 adjusted dynamically depending on the traffic conditions. More jitter
 = larger buffer to try to compensate for late packets rather than
 considering them lost. Anyone correct me if I'm wrong here.

 Having said that, I don't notice the delay or distorted voice even if
 the decode latency is as high as 80 ms. Not sure about 200+ ms, but it
 seems rather high and would imply to me that you have a connectivity
 issue somewhere on your LAN.

The explanation of jitter adding to decode latency sounds reasonable.
However, as I said before, I have never seen jitter go above 5ms even
when our decode latency spirals out of control.

Our latency is under 1ms, generally. It's 100 base T fully switched, and
not highly utilized, with 2 switches between the phone and the PBX.

Our current working theory, which we will test soon, is that this may
be caused by periodic high levels of ARP broadcast traffic. I'm not
familiar with the hardware of these phones, and for most ethernet
devices they should ignore ARP with no performance effects. But if the
SPA-841 is set up in such a way that it eats CPU for the phone to
discard ARP packets, then this could be a problem for us.

I'll keep you posted on what we find. If anyone has any insight into the
networking hardware the SPA-841 uses, I'd be interested in that.

Thanks,

Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]
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[Asterisk-Users] Compiling SpanDSP

Has anybody been successful with compiling the pre3 version of SpanDSP 
on the current Asterisk CVS?  I'm getting:


app_rxfax.c: In function `phase_e_handler':
app_rxfax.c:77: warning: implicit declaration of function 
`fax_get_transfer_statistics'
app_rxfax.c:78: warning: implicit declaration of function 
`fax_get_far_ident'
app_rxfax.c:79: warning: implicit declaration of function 
`fax_get_local_ident'

app_rxfax.c:93: error: structure has no member named `callerid'
app_rxfax.c: In function `rxfax_exec':
app_rxfax.c:263: warning: passing arg 1 of `fax_init' from incompatible 
pointer type

app_rxfax.c:264: error: structure has no member named `verbose'
app_rxfax.c:267: warning: implicit declaration of function 
`fax_set_local_ident'
app_rxfax.c:270: warning: implicit declaration of function 
`fax_set_header_info'

app_rxfax.c:271: warning: implicit declaration of function `fax_set_rx_file'
app_rxfax.c:273: warning: implicit declaration of function 
`fax_set_phase_d_handler'
app_rxfax.c:274: warning: implicit declaration of function 
`fax_set_phase_e_handler'

app_rxfax.c:285: warning: implicit declaration of function `fax_rx_process'
app_rxfax.c:288: warning: implicit declaration of function `fax_tx_process'
app_rxfax.c:325: warning: passing arg 1 of `fax_release' from 
incompatible pointer type

app_rxfax.c: At top level:
app_rxfax.c:61: warning: 't30_flush' defined but not used
make[1]: *** [app_rxfax.o] Error 1
make[1]: Leaving directory `/home/doug/cvs/10032005/asterisk/apps'
make: *** [subdirs] Error 1

Doug

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Re: [Asterisk-Users] Nufone

2005-10-03 Thread Tom Vile

Where is it getting the extra 8 from? It seems like
you are passing an invalid number to the trunk.
Spawn extension (crystal-sip, 8800759, 1)On 10/3/05, Crystal Stream, Incorporated
 [EMAIL PROTECTED] wrote:
crystalstream*CLI-- Executing Macro(SIP/3044-5300,outvoip-2|1800759) in new stack-- Executing SetCIDName(SIP/3044-5300, CRYSTALSTREAM NET|a) in new st ack
-- Executing SetCIDNum(SIP/3044-5300,866xxx|a) in new stack-- Executing Authenticate(SIP/3044-5300,123987) in new stack-- Playing 'agent-pass' (language 'en')
-- Playing 'auth-thankyou' (language 'en')-- Executing Monitor(SIP/3044-5300, wav) innew stack-- Executing Ringing(SIP/3044-5300, ) in newstack
-- Executing Wait(SIP/3044-5300, 2) in newstack-- Executing Dial(SIP/3044-5300,IAX2/[EMAIL PROTECTED]/1800759 ) in newstack-- Called 
[EMAIL PROTECTED]/1800759Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type:IAX Subclass: NEW Timestamp: 3msSCall: 1DCall: 0[66.225.202.72:4569
] VERSION : 2 CALLED NUMBER : 1800759 CALLING NUMBER: 8663113060 LANGUAGE: en USERNAME: username-hidden FORMAT: 4 CAPABILITY: 63502
 ADSICPE : 2 DATE TIME : 188966086Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type:IAX Subclass: AUTHREQ Timestamp: 3msSCall: 00061DCall: 1[
66.225.202.72:4569] AUTHMETHODS : 2 CHALLENGE : 150617580 USERNAME: username-hiddenTx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type:IAX Subclass: AUTHREP Timestamp: 00033msSCall: 1DCall: 00061
[66.225.202.72:4569] MD5 RESULT: c8214533976d4dec8b233543dac0eaacRx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type:IAX Subclass: ACCEPT Timestamp: 00036msSCall: 00061DCall: 1
[66.225.202.72:4569] FORMAT: 4-- Call accepted by 66.225.202.72 (format ulaw)-- Format for call is ulaw
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type:IAX Subclass: ACK Timestamp: 00036msSCall: 1DCall: 00061[66.225.202.72:4569]Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type:
VOICE Subclass: 4 Timestamp: 00060msSCall: 1DCall: 00061[66.225.202.72:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type:VOICE Subclass: 4
 Timestamp: 00080msSCall: 00061DCall: 1[66.225.202.72:4569]Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type:IAX Subclass: ACK Timestamp: 00080msSCall: 1DCall: 00061
[66.225.202.72:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type:IAX Subclass: ACK Timestamp: 00060msSCall: 00061DCall: 1[
66.225.202.72:4569]Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type:CONTROL Subclass: (15?) Timestamp: 00123msSCall: 00061DCall: 1[66.225.202.72:4569
]Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type:IAX Subclass: ACK Timestamp: 00123msSCall: 1DCall: 00061[66.225.202.72:4569]Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 003 Type:
CONTROL Subclass: (14?) Timestamp: 01423msSCall: 00061DCall: 1[66.225.202.72:4569]Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type:IAX Subclass: ACK
 Timestamp: 01423msSCall: 1DCall: 00061[66.225.202.72:4569]-- IAX2/NuFone/1 is making progress passing it toSIP/3044-5300Oct3 12:38:19 WARNING[49584]: app_dial.c:372
wait_for_answer: Unable to forwar d frame-- Hungup 'IAX2/NuFone/1'== Spawn extension (macro-outvoip-2, s, 7) exitednon-zero on 'SIP/3044-5300' in macro 'outvoip-2'== Spawn extension (crystal-sip, 8800759, 1)
exited non-zero on 'SIP/3044- 5300'--- Tom Vile [EMAIL PROTECTED] wrote: how many digits is your callerid passing to the trunk? I am seeing 11
 8663xx3 is that correct? I had an issue last week with passing to many digits to my provider and the call would hang up immediately. You could also turn debugging on for this so we can
 get a better log. iax2 debug peer nufone On 10/3/05, Crystal Stream, Incorporated [EMAIL PROTECTED] wrote:
   After -- IAX2/NuFone/3 is making progress passing it  to SIP/3044-bcd0 I'm getting a Busy tone and it's  not even connecting the call.
     -- Executing Macro(SIP/3044-bcd0,  outvoip-2|1800759) in new stack  -- Executing SetCIDName(SIP/3044-bcd0, X X
 X|a)  in new stack  -- Executing SetCIDNum(SIP/3044-bcd0,  8663xx3|a) in new stack  -- Executing Authenticate(SIP/3044-bcd0,
 xx)  in new stack  -- Playing 'agent-pass' (language 'en')  -- Playing 'auth-thankyou' (language 'en')  -- Executing Monitor(SIP/3044-bcd0, wav) in
 new  stack  -- Executing Ringing(SIP/3044-bcd0, ) in new stack  -- Executing Wait(SIP/3044-bcd0, 2) in new stack
  -- Executing Dial(SIP/3044-bcd0,  IAX2/[EMAIL PROTECTED]/1800759) in new stack  -- Called [EMAIL PROTECTED]/1800759  -- Call accepted by 
66.225.202.72 http://66.225.202.72 (format ulaw)  -- Format for call is ulaw  -- IAX2/NuFone/3 is making progress passing it to  SIP/3044-bcd0
  -- Hungup 'IAX2/NuFone/3'  == Spawn extension (macro-outvoip-2, s, 7) exited  non-zero on 'SIP/3044-bcd0' in macro 'outvoip-2'  == Spawn extension (crystal-sip, 8800759, 1)
  exited non-zero on 'SIP/3044-bcd0'   x*CLI iax2 show peers  Name/Username Host Mask  Port Status  voicepulse2/Fbg 
66.234.228.166

[Asterisk-Users] Console sound output -- shuts off when call from console answered

2005-10-03 Thread Wolfgang Borgon

I've got a problem with audio output from the Asterisk console. I'd really appreciate any help.

I'm simply trying to dial out to a phone on PSTN. My extensions.conf entry is as follows:

exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})exten = _1NXXNXX,2,HangupAfter starting asterisk and dialing, I hear a ringback tone through the console speaker, and the PSTN phone rings.

1) If I answer the phone, asterisk indicates the call has been answered, says something about stopping sounds, and the console speaker cuts out. I am only able to speak through the console mic and hear this speech on the PSTN phone. After hanging up, if I try dialing again, I no longer hear any ringback tones -- basically it seems that the console speaker has been shut off.

2) If I don't answer the phone, and hangup the call from the console, I can continue to dial out and here ringback output on the console speaker.

Please, I'd really appreciate any advice. I'm assuming this is an Asterisk issue has I've had no other problems with my ALSA/JACK config.

Below is sample verbose output from the console:

___*CLI dial [EMAIL PROTECTED] -- Executing Dial("ALSA/default", IAX2/[EMAIL PROTECTED]/1###|) in new stack -- Called [EMAIL PROTECTED]/1## -- Call accepted by 66.246.246.52 (format ulaw) -- Format for call is ulaw -- IAX2/voxee-1 is making progress passing it to ALSA/defaultOct 2 14:44:04 WARNING[3750]: chan_alsa.c:751 alsa_indicate: Don't know how to display condition 14 on ALSA/default
-- IAX2/voxee-1 is ringing -- IAX2/voxee-1 stopped sounds -- IAX2/voxee-1 answered ALSA/default Console call has been answered -- Hungup 'IAX2/voxee-1' == Spawn extension (voxee, 1##, 3) exited non-zero on 'ALSA/default' Hangup on console ___CLI
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Re: [Asterisk-Users] Compiling SpanDSP

2005-10-03 Thread Dave Cotton

On Mon, 2005-10-03 at 14:10 -0400, Doug Lytle wrote:
 Has anybody been successful with compiling the pre3 version of SpanDSP 
 on the current Asterisk CVS?  I'm getting:
 
 app_rxfax.c: In function `phase_e_handler':
 app_rxfax.c:77: warning: implicit declaration of function 
 `fax_get_transfer_statistics'
 app_rxfax.c:78: warning: implicit declaration of function 
 `fax_get_far_ident'
 app_rxfax.c:79: warning: implicit declaration of function 
 `fax_get_local_ident'
 app_rxfax.c:93: error: structure has no member named `callerid'

Look at rxfax.c around line 88 there's an #if statement remove the
references to callerid.

This error has been around for a while.


-- 
Dave Cotton [EMAIL PROTECTED]

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[Asterisk-Users] Asterisk 1.0.8 and TDM stop acking inbound calls?

2005-10-03 Thread Patrick Friedel

My office has been running Asterisk 1.0.8 and a TDM04B for a few months 
now without too much trouble.  After a while we discovered that after a 
certain period (about a month), asterisk stopped acknowledging inbound 
calls.  Since I was out of the office the first time it happened, 
another admin rebooted the whole box which solved the problem.  The 
second time it happened I discovered that just restarting gracefully 
solved the problem, so I put that into my cron and forgot about it.  (I 
know, it's not right, but debugging something that happens 
unpredictably once a month could go on for way too long to be acceptable..)


 Now, less than a week since I did that, asterisk stopped ringing our 
extensions on inbound calls again.  sip show peers showed that 
asterisk knew about all of the extensions.  I forgot to check zap show 
channels when * was ignoring inbound calls, is it possible that * 
thinks all the lines are still off hook?  Is there anything else I 
should do to figure out what's causing trouble?  Unfortunately it's 
usually something of a panic situation, so I'm not allowed the chance to 
troubleshoot as thoroughly as I'd like.


 Speaking of, I've fiddled and tweaked left and right to get hangup 
detection working better to no real avail.  Asterisk eventually decides 
the far side hung up about 10 seconds after the fact.  Am I 
understanding right that call progress is still something of a black art 
for analog FXO devices?  Not getting 10 second dead air voicemails when 
people hang up would be sweet. :)


 I couldn't find a changelog for 1.0.9 to see if it's worth the 
off-hours maintenance window, and we're too dependant on the phones to 
try 1.2.  Should I try the next step up in the probably unnecessary 
preventative maintenance and unload/reload the wctdm module during the 
asterisk restart?  Is there any way to have asterisk notify that it's 
running low on/out of resources?  We don't typically ever tie up all of 
our zap channels except for really particularly exciting days, so if 
they are all in use it would be cause for concern..

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Re: [Asterisk-Users] Compiling SpanDSP


Dave Cotton wrote:


On Mon, 2005-10-03 at 14:10 -0400, Doug Lytle wrote:
 


Look at rxfax.c around line 88 there's an #if statement remove the
references to callerid.

This error has been around for a while.

 




That took care of the callerid compile error, but not the verbose error:

error: structure has no member named `verbose'

Doug

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Re: [Asterisk-Users] Compiling SpanDSP

2005-10-03 Thread Tom Hayden

I've been getting the same problem with the verbose issue. I just
commented out the line, and it seemed to compile OK.

--
Tom

On 10/3/05, Doug Lytle [EMAIL PROTECTED] wrote:
 Dave Cotton wrote:

 On Mon, 2005-10-03 at 14:10 -0400, Doug Lytle wrote:
 
 
 Look at rxfax.c around line 88 there's an #if statement remove the
 references to callerid.
 
 This error has been around for a while.
 
 
 


 That took care of the callerid compile error, but not the verbose error:

 error: structure has no member named `verbose'

 Doug

 --

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 liberty nor safety.


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--
Tom
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Re: [Asterisk-Users] Adding Cepstral to Asterisk

2005-10-03 Thread Matthew Gibson


Wojciech Tryc wrote:


I am not following...
Why would you need to integrate Cepstral directly into Asterisk? Just 
to be able to call it as Asterisk app from your dialplan? I am running 
Cepstral and calling it through the System call.



You could try the howto located here:

http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt

for cepstral integration into asterisk. It makes it app_cepstral, instead
of using system calls.

Mat

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Re: [Asterisk-Users] Asterisk 1.0.8 and TDM stop acking inbound calls?

2005-10-03 Thread Patrick Friedel


Patrick Friedel wrote:

I couldn't find a changelog for 1.0.9 to see if it's worth the 
off-hours maintenance window, and we're too dependant on the phones to 
try 1.2.  Should I try the next step up in the probably unnecessary 
preventative maintenance and unload/reload the wctdm module during the 
asterisk restart?  Is there any way to have asterisk notify that it's 
running low on/out of resources?  We don't typically ever tie up all 
of our zap channels except for really particularly exciting days, so 
if they are all in use it would be cause for concern..



 In the interim, and completely on a whim, I've put a couple of 
splitters and added another FXO device onto the line, a good old 
fashioned analog phone that chirps once until asterisk picks up the 
line.  With any luck, the next time asterisk takes a dive, that phone 
will continue to ring and we'll catch on faster than the first customer 
that decides to email us wondering why nobody is picking up. :)


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[Asterisk-Users] Real Life FAX sending receiving

2005-10-03 Thread Jenna Cole

is it possible to achive the following scenario?

faxmachine--tdm40bFXS--SIPnetwork--Gateway--faxmachine

i have found a lot of documents about asterisk
receiving a fax and saving it to a file. But i want to
receive the fax via SIP and send it to my faxmachine.
I also want to send a fax from my faxmachine through
the digium card, so asterisk should send the fax via
SIP to the gateway, which also has a faxmachine
connected.

is this possible?
would anyone be so kind to send me the config files or
config tips?

thanks a lot guys ;)
Jenna.







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[Asterisk-Users] Problem with configuration of Quintum AX with Asterisk

2005-10-03 Thread Michal Misiak

Hi. 
I'm trying to configurate Quintum AX to work with Asterisk SIP egister/Proxy
server and my problem is that only the first user account get logged in and
only that user is able to make call correctly. It seems to be a problem with
authorization - I have noticed no Proxy-Authorization information in SIP
INVITE, ACK, CANCEL messages. I have also noticed that when I remove secret
from Asterisk sip.conf file (no authorization required) other users
(accounts) can make a call but no media are sent. 

Do you know reasons of this problem and can you help me resolving it.

Michał Misiak
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Re: [Asterisk-Users] Real Life FAX sending receiving


Jenna Cole wrote:


receive the fax via SIP and send it to my faxmachine.
I also want to send a fax from my faxmachine through
the digium card, so asterisk should send the fax via
SIP to the gateway, which also has a faxmachine
connected.

is this possible?
 


Short answer, no.  Long answer can be found here:

http://www.soft-switch.org/spandsp_faq/ar01s04.html

Doug

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nor safety.


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Re: [Asterisk-Users] Asterisk 1.0.8 and TDM stop acking inbound calls?


 My office has been running Asterisk 1.0.8 and a TDM04B for a few months 
 now without too much trouble.  After a while we discovered that after a 
 certain period (about a month), asterisk stopped acknowledging inbound 
 calls.  Since I was out of the office the first time it happened, 
 another admin rebooted the whole box which solved the problem.  The 
 second time it happened I discovered that just restarting gracefully 
 solved the problem, so I put that into my cron and forgot about it.  (I 
 know, it's not right, but debugging something that happens 
 unpredictably once a month could go on for way too long to be acceptable..)
 
   Now, less than a week since I did that, asterisk stopped ringing our 
 extensions on inbound calls again.  sip show peers showed that 
 asterisk knew about all of the extensions.  I forgot to check zap show 
 channels when * was ignoring inbound calls, is it possible that * 
 thinks all the lines are still off hook?  Is there anything else I 
 should do to figure out what's causing trouble?  Unfortunately it's 
 usually something of a panic situation, so I'm not allowed the chance to 
 troubleshoot as thoroughly as I'd like.
 
   Speaking of, I've fiddled and tweaked left and right to get hangup 
 detection working better to no real avail.  Asterisk eventually decides 
 the far side hung up about 10 seconds after the fact.  Am I 
 understanding right that call progress is still something of a black art 
 for analog FXO devices?  Not getting 10 second dead air voicemails when 
 people hang up would be sweet. :)
 
   I couldn't find a changelog for 1.0.9 to see if it's worth the 
 off-hours maintenance window, and we're too dependant on the phones to 
 try 1.2.  Should I try the next step up in the probably unnecessary 
 preventative maintenance and unload/reload the wctdm module during the 
 asterisk restart?  Is there any way to have asterisk notify that it's 
 running low on/out of resources?  We don't typically ever tie up all of 
 our zap channels except for really particularly exciting days, so if 
 they are all in use it would be cause for concern..

Check the revision of the TDM card. If rev E/F, call digium support to
get it replaced. Known problem with early versions of the card.


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[Asterisk-Users] Asterisk Ignoring [User] in SIP.CONF

2005-10-03 Thread Andre Sharpe


Hi,


I have a SIP.CONF with a user section like this:-

[1234]
accountcode=HABITAZ
type=friend
callerid=HABITAZ/1234
context=milkshake
userName=1234
secret=1234
host=dynamic
dtmfmode=rfc2833
qualify=yes
callgroup=1
pickupgroup=1
canreinvite=no


When I login from a X-Lite phone, with Username, Authentication User and 
Password set to '1234', everything works fine. The
account code is set to 'HABITAZ' and everything seems fine.

But when the Username on X-Lite is changed to pretty much anything else, eg. 
'4321' - and the Authentication User and Password
left to be '1234', Asterisk still allows calls to be made from the phone, but 
the CDR countains a blank accountcode.

How can I disallow login if the Username field on X-Lite is not set to a 
valid user section in SIP.conf?

Andre



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Re: [Asterisk-Users] Adding Cepstral to Asterisk

2005-10-03 Thread pbx

the app_cepstral.c file had a problem that it was trying use

#include ../asterisk.h

I had to force it to where asterisk.h was located... in my case it was in
/usr/src/asterisk/include
so i changed the #include to say

#include /usr/src/asterisk/include/asterisk.h and then it would compile
through with no problems



 Wojciech Tryc wrote:

 I am not following...
 Why would you need to integrate Cepstral directly into Asterisk? Just
 to be able to call it as Asterisk app from your dialplan? I am running
 Cepstral and calling it through the System call.

 You could try the howto located here:

 http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt

 for cepstral integration into asterisk. It makes it app_cepstral, instead
 of using system calls.

 Mat

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Re: [Asterisk-Users] SIP 400 Bad Request from Cisco 7960/7940

2005-10-03 Thread Corey S. McFadden



Olle,

Thanks for looking into it.  In doing some ngrep work I figured out where 
my problem is.

Acutal error from the 79xx inside the SIP header is:
 Warning: 399 Bad Request - 'Malformed/Missing FROM: field'

From looks like this:
 From: Sales Queue sip:12345...

Those double-quotes looked bad, so I assumed that the problem was related 
to this:  Set(CALLERID(name)=Sales Queue) that executes before the 
offending queue.

I changed to: Set(CALLERID(name)=Sales) and no success.
then to: Set(CALLERID(name)=Sales) and it's OK.

Am I just using the Set() command wrong?  It seems pretty 
counter-intuitive not to enclose multi-word strings in quotes but if 
that's the problem let me know.


FYI, we're testing with (right now) CVS-Nv1-2-0-beta1-10/01/05-20:43:03

SIP Firmware on the phone is 7.4.

-Corey




On Sun, 2 Oct 2005, Olle E. Johansson wrote:

 Doug Lytle wrote:
  Olle E. Johansson wrote:
  
  Corey S. McFadden wrote:
   
 
  Here's the CLI output:
 -- Got SIP response 400 Bad Request back from 192.168.249.94
 -- SIP/502-9a58 is circuit-busy
 
  I've tried a few different Asterisk versions CVS-HEAD, stable, even
  1.2 beta.  I've also bounced between SIP firmware 7.4 and 7.5 on the
  7960/7940 phones.
  
  As of Friday evening, we've been seeing this on our system as well. 
  Olle, do you want debugs from other people as well, or will the one
  you've requested be enough?
 Just make sure I get one. If I can't figure that one out, I might need
 more. Thank you for asking. The first one in my mailbox tomorrow morning
 (it's late in Sweden) will get my attention :-)
 
 I need to know version of Asterisk as well.
 
 As we are getting very close to release, it's important for us to track
 down and resolve all outstanding bugs as quickly as possible. The SIP
 channel has been changing quite a lot during the last two months, so
 there are a lot of new code in there right now.
 
 Thank you for your assistance!
 
 /O
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 dangerous content, and is believed to be clean.
 


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[Asterisk-Users] sip phones on x86_64

2005-10-03 Thread Wayne Gemmell

Hi all

Can anyone recommend a good soft phone that can compile on x86_64 (linux) 
platform?


-- 
Regards

Wayne Gemmell

Tel  Fax: (011) 894-4081
Cell  : 072 836 4325
Email  : [EMAIL PROTECTED]
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Re: [Asterisk-Users] Adding Cepstral to Asterisk

2005-10-03 Thread trixter http://www.0xdecafbad.com

On Mon, 2005-10-03 at 13:02 -0700, [EMAIL PROTECTED] wrote:
 the app_cepstral.c file had a problem that it was trying use
 
 #include ../asterisk.h
 
 I had to force it to where asterisk.h was located... in my case it was in
 /usr/src/asterisk/include
 so i changed the #include to say
 
 #include /usr/src/asterisk/include/asterisk.h and then it would compile
 through with no problems
 


try adding -I/path/to/asterisk/includes  in your case
-I/usr/src/asterisk/include to your cc/gcc line (in Makefile usually
CCOPTS var, but I havent looked at that Makefile specifically).

This is the more elegant solution :P   Or install your asterisk includes
in the system default (/usr/include normally)

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] SIP 400 Bad Request from Cisco 7960/7940

2005-10-03 Thread steve



On Mon, 3 Oct 2005, Corey S. McFadden wrote:

 Am I just using the Set() command wrong?  It seems pretty 
 counter-intuitive not to enclose multi-word strings in quotes but if 
 that's the problem let me know.

Yeah, that's the problem.

Steve

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RE: [Asterisk-Users] SIP-CPE Gateway

2005-10-03 Thread Anders Svensson

Just to clarify. These products are not produced by this company, its
Taiwanese brands. The SIP-CPE Gateway is a rebranded VodTel MOSA 3700

Anders

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
Michaelson
Sent: den 3 oktober 2005 18:12
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP-CPE Gateway

Has anyone used the GSM-SIP gateway product produced by a company at 
sipcpe.com?  Any comments?


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Re: [Asterisk-Users] What does the error stale nonce' mean?

2005-10-03 Thread Morten Isaksen


On 10/3/05, Olle E. Johansson [EMAIL PROTECTED] wrote:
 Does anyone know what "stale nonce" is?I've answered this question many times, so you should be able to find
the answer...A stale nonce is when a device tries to re-authenticate with a noncethat is no longer valid. We are telling them that the nonce they used isinvalid, and re-issue a new challenge and a fresh nonce. It's just an
informative message, that I propably should move away to a debug levelof some kind.


I get this error when I use a Audiocodes MP-124 against Asterisk 1.2beta1 and asterisk refuses the call. When I useCVS-D2005.02.12.14.37.11-04/13/05-16:14:03 it works fine.

I do not have access to the debug and log file now, but I will send them tomorrow.

/Morten

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Re: [Asterisk-Users] sip phones on x86_64

2005-10-03 Thread Dennis Gilmore

Once upon a time Monday 03 October 2005 3:11 pm, Wayne Gemmell wrote:
 Hi all

 Can anyone recommend a good soft phone that can compile on x86_64 (linux)
 platform?
kphone compiles and is available in Fedora extras  and im sure is available 
for other distros.  If you want to get adventurous you could try cvs 
gnomemeeting.  it also has sip support.

Dennsi


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Re: [Asterisk-Users] Asterisk on windows

2005-10-03 Thread Christopher Dobbs


Matt wrote:


Extremely good point... I myself am a Linux person, but manage several
Windows machines (several meaning 25 or so).   There is definately a
time and place for Windows.. I'm just not sure a real-time-VoIP server
is the time or place.Being semi-half serious about the GUI there
also.You install X on your Asterisk server and things will not be
happy either.
 

I Run SuSE 9.3 with KDE 3.4, Asterisk 1.0.3, play MP3's and OGG's, SAMBA 
services, HTTPD, VNC, MicroWindows, FTP, SMTP, POP, IMAP, plus others.
I dont see that the GUI slows things down to much, unless I am running a 
test and gring the call volume over 500 active calls. (I am developing a 
new channel driver for * ment for inclusion in mobile phones, think 
Asterisk+Cell Phone).  The assertion that a GUI will bring a system to 
it's knee's is utter CRAP!  It all has to do whith what the system is 
doing besides, and what the hardware can handle. BTW: the system this 
all is running on is an AMD 1700+, and the same system that I am using 
to brows the mailing list.


--Christopher Dobbs
--I think I think, There for I think I am.

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Re: [Asterisk-Users] What does the error stale nonce' mean?

2005-10-03 Thread trixter http://www.0xdecafbad.com

A stale nonce is more of a warning than an error.  In SIP your
authorization credentials are encoded in the SIP headers.  To prevent
people from capturing that data and using it later to make calls on your
account a nonce is used.

A nonce is a disposable number that is added to the string a hash
algorithm will hash.  This makes hashing algorithms (like md5) have
different output.  This is a common cryptography technique.  

The SIP RFC requires that the nonce randomly change periodically.  If
the client uses a nonce that was expired it is considered a 'stale
nonce'.  The client should then get the current nonce and use that
instead.  This message lets you know that the client tried to use a
stale nonce, which can indicate someone trying a replay attack (using
captured data from a previous session) or a client that isnt properly
getting the new nonce, or even just timing issues as follows:

Client gets a nonce.  
Client goes to register/reregister using that nonce
At the same time the client is preparing the message to 
 register/reregister the server chooses a new nonce
Client sends the message with the now old nonce

Then again it could be something else entirely :)


On Mon, 2005-10-03 at 22:35 +0200, Morten Isaksen wrote:
 
 On 10/3/05, Olle E. Johansson [EMAIL PROTECTED] wrote: 
  Does anyone know what stale nonce is?
 I've answered this question many times, so you should be able
 to find 
 the answer...
 
 A stale nonce is when a device tries to re-authenticate with a
 nonce
 that is no longer valid. We are telling them that the nonce
 they used is
 invalid, and re-issue a new challenge and a fresh nonce. It's
 just an 
 informative message, that I propably should move away to a
 debug level
 of some kind.
  
  
 I get this error when I use a Audiocodes MP-124 against Asterisk
 1.2beta1 and asterisk refuses the call. When I
 use CVS-D2005.02.12.14.37.11-04/13/05-16:14:03 it works fine.
  
 I do not have access to the debug and log file now, but I will send
 them tomorrow.
  
 /Morten
  
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Re: [Asterisk-Users] SIP 400 Bad Request from Cisco 7960/7940